TOS QoS Values Not Changing

FreePBX
1

The following values are present in sip_general_additional.conf in the latest stable FreePBX 14 Asterisk 13 release.

tos_sip=cs3
tos_audio=ef
tos_video=af41

Adding new values to Settings–>SIP Settings–>Chan-SIP does not seem to override these values. Also, a wireshark analysis of the UDP packets reveals they are being sent with a DSCP tag of 0x05. I cannot match this to a known hex value to then translate into a TOS value.

How would one go about properly changing these values assuming that there isn’t a bug preventing this?

2

my understanding (someone please correct me if I am wrong) is that on the distro, freepbx does not run as root. as a result, these values cannot be changed because root is the only one that is allowed to set qos…I would like to know if there is a way, but I don’t think so.

what I am doing is using the switch to mark the ip of the pbx with the qos I want.

good luck

3

My current router can only pass DSCP values between the various ports/subnets. It cannot insert/modify these parameters.

It’s current QoS works great for traffic leaving my WAN but I’m trying to tweak internal traffic. My standalone WiFi APs respect the DSCP tags and actually intervene to prioritize those packets.

It’s made a huge difference on the sending side from people’s smartphones here since some of the softphone apps I’ve deployed can set DSCP tags.

4

yes, i wish there was a way for freepbx (asterisk) to do this. I have looked around and I don’t see a way. I was hoping someone else would chime in here with a brilliant idea.

good luck

5

I can confirm that you CAN add tos_* values to the bottom of the Chan_SIP settings tab, and redefine the vales from FreePBX/Asterisk defaults. You can confirm the settings have been applied with:

asterisk -x "sip show settings" | grep -A10 "QoS"

You must set them to a value recognized by Asterisk as outlined here:
https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service

6

I followed those exact guidlines and a wirehshark analysis showed a DSCP of 0x05 instead of ef or 46.

Note: tos_*=ef is equivalent to a DSCP of 46.

Also, I do see packets from my endpoints (both hard phones and soft) making it to the PBX with tos_sip=ef tags.

7

Lorne, i defer to your knowledge, but all the post about that say the it does not matter unless you run asterisk as root…which freepbx does not do…am i wrong?